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Add received audio/video call duration metrics based on packets.

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Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.

BUG=webrtc:7882

Review-Url: https://codereview.webrtc.org/2957073002
Cr-Commit-Position: refs/heads/master@{#18881}
This commit is contained in:
saza 2017-07-04 00:19:22 -07:00 committed by Commit Bot
parent 2a2b297aa6
commit 746749237a
2 changed files with 26 additions and 0 deletions
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24
webrtc/call/call.cc
View File

@ -325,6 +325,10 @@ class Call : public webrtc::Call,
RateCounter received_audio_bytes_per_second_counter_;
RateCounter received_video_bytes_per_second_counter_;
RateCounter received_rtcp_bytes_per_second_counter_;
rtc::Optional<int64_t> first_received_rtp_audio_ms_;
rtc::Optional<int64_t> last_received_rtp_audio_ms_;
rtc::Optional<int64_t> first_received_rtp_video_ms_;
rtc::Optional<int64_t> last_received_rtp_video_ms_;
// TODO(holmer): Remove this lock once BitrateController no longer calls
// OnNetworkChanged from multiple threads.
@ -530,6 +534,16 @@ void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) {
}
void Call::UpdateReceiveHistograms() {
if (first_received_rtp_audio_ms_) {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds",
(*last_received_rtp_audio_ms_ - *first_received_rtp_audio_ms_) / 1000);
}
if (first_received_rtp_video_ms_) {
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds",
(*last_received_rtp_video_ms_ - *first_received_rtp_video_ms_) / 1000);
}
const int kMinRequiredPeriodicSamples = 5;
AggregatedStats video_bytes_per_sec =
received_video_bytes_per_second_counter_.GetStats();
@ -1317,6 +1331,11 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
event_log_->LogRtpHeader(kIncomingPacket, packet, length);
const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
if (!first_received_rtp_audio_ms_) {
first_received_rtp_audio_ms_.emplace(arrival_time_ms);
}
last_received_rtp_audio_ms_.emplace(arrival_time_ms);
return DELIVERY_OK;
}
} else if (media_type == MediaType::VIDEO) {
@ -1324,6 +1343,11 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
received_bytes_per_second_counter_.Add(static_cast<int>(length));
received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
event_log_->LogRtpHeader(kIncomingPacket, packet, length);
const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
if (!first_received_rtp_video_ms_) {
first_received_rtp_video_ms_.emplace(arrival_time_ms);
}
last_received_rtp_video_ms_.emplace(arrival_time_ms);
return DELIVERY_OK;
}
}

2
webrtc/video/end_to_end_tests.cc
View File

@ -2644,6 +2644,8 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx,
// Verify that stats have been updated once.
EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
EXPECT_EQ(1, metrics::NumSamples(
"WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));